[asterisk-users] Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet

[michel freiha]

Maybe you have a Codec issue?

On Tue, Oct 28, 2008 at 11:20 PM, Benny Amorsen
<benny+usenet at amorsen.dk<benny%2Busenet at amorsen.dk>
> wrote:

> Lincoln King-Cliby <lincoln at controlworks.com> writes:
>
> > Periodically I'm seeing calls placed from the 7961s through anything
> > on the PBX that requires digit entry (the Auto Attendant, Voicemail,
> > etc.) 'randomly' drop; extension-to-extension calls
> > extension-to-PSTN, and PSTN-to-extension calls never have any issues
> > whatsoever. Nor have I been able to duplicate the issues hopping
> > around auto attendants on an inbound PSTN call.
>
> I am not sure this is relevant in the 1.4.x versions, but here goes
> anyway:
>
> In Asterisk 1.2.x it could sometimes happen that Asterisk believed the
> path to a server was so good, that it would only allow 1 ms for
> answers to be received. It would do all its retransmissions in less
> than 200ms, and then it would complain about no reply to critical
> packet.
>
> Anyway, you can adjust the minimum timer with the configuration option
> t1min in sip.conf. I would recommend setting it to at least 100 (it is
> in ms) and perhaps 500 would help for you.
>
> It is also highly possible that your issue is completely different.
>
>
> /Benny
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081029/011915b1/attachment.htm